(a) Field of the Invention:
This invention relates to a digital signal processing device of which the whole structure is simplified by adding, before digital-to-analog conversion processing, interpolation data obtained by using the polynominal interpolation theorem at an intermediate time point between sample data.
(b) Description of the Prior Art:
In the digital audio technique, when the original audio signal is to be obtained through demodulation of its digital signal, the digital signal is converted into analog signal by a digital-to-analog converter (D-A converter) 1 and then subjected to processings through a buffer amplifier 2, low-pass filter 3, and a buffer amplifier 4 as shown in FIG. 1. The low-pass filter 3 is provided to remove the harmonic components from the output signal of the D-A converter 1. As shown in FIG. 2, these harmonic components are formed as frequency components of the original signal are generated on both sides of frequencies which are integer multiples of the sampling frequency f.sub.s. Since these harmonic components comprise frequency components located near the upper limit of the band of the original signal, the low-pass filter 3 is required to have steep cutoff characteristics. For example, in the case of compact discs where the band of the original signal is set at a range of 0-20 kHz (the sampling frequency being 44.1 kHz), such steep cutoff characteristics as to attain an attenuation rate of .+-.1 dB at 0-20 kHz and -90 dB at 24 kHz and over is required. Therefore, a Chebychev-type low-pass filter, which possesses such steep cutoff characteristics, is generally used. However, improvement in the cutoff characteristics of the low-pass filter results in such disadvantages as expensive production cost due to necessity for providing a filter of a higher order, deterioration of sound quality due to an increased number of elements, and substantial waveform distortion caused by considerable phase change near the upper limit of the passband.
Thus attempts have been made to simplify the structure of the low-pass filter 3 by providing a digital filter 5 as shown in FIG. 3 to cut off the harmonic components generated near the upper limit of the band of the original signal before effecting digital-to-analog conversion, thereby to lessen the burden of the low-pass filter 3 disposed in the posterior stage.
However, the conventional digital filter has a long data bit length and requires a high-speed multiplier and adder as well as a RAM and ROM, resulting in complexity of the hardware structure and hence high production cost.